Method and system for providing interdomain traversal in support of packetized voice transmissions

ABSTRACT

An approach provides interdomain traversal to support packetized voice transmissions. A request for establishing a voice call is received from a source endpoint behind a first network address translator of a first domain, wherein the request specifies a directory number of a destination endpoint within a second domain. A network address is determined for communicating with the destination endpoint based on the directory number. Additionally, existence of a second network address translator within the second domain is determined. If the network address can be determined, a media path is established between the source endpoint and the destination endpoint based on the network address to support the voice call.

RELATED APPLICATIONS

This application is related to, and claims the benefit of the earlierfiling date under 35 U.S.C. § 119(e) of, U.S. Provisional PatentApplication (Ser. No. 60/601,256), filed Aug. 13, 2004, entitled“Communications System with SIP-based Fixed-Mobile Convergence,” andU.S. Provisional Patent Application (Ser. No. 60/606,605), filed Sep. 2,2004, entitled “IP Interconnect Architecture”; the entireties of whichare incorporated herein by reference.

FIELD OF THE INVENTION

The present invention, according to various embodiments, relates tocommunications, and more particularly, to transmitting a packetizedvoice call across different domains.

BACKGROUND OF THE INVENTION

Internet Protocol (IP) telephony has changed the business model andengineering approaches of how voice services are provisioned anddelivered. The attractive economics of IP telephony (stemming largelyfrom the global connectivity and accessibility of the Internet) alongwith innovative productivity tools for users have triggered adoption ofthis technology by numerous businesses, organizations, enterprises andthe like. Unfortunately, this adoption primarily has been uncoordinated,and driven by the needs of the specific enterprise little regard to a“global” approach for IP telephony deployment. Interestingly, theprevailing IP telephony implementations have confined the particularenterprises, as to make communications outside the enterprise difficultand impractical.

As enterprises implement Internet telephony as well as messaging systemsand associated applications, closed communities of IP enabled users arecreated—i.e., “IP islands”. That is, because of systems and applicationsconstraints and incompatibilities, these IP enable users are isolated,and thus, cannot readily communicate with each other. Moreover, asInternet Service Providers (ISPs), cable, and mobile network operatorsbegin to provide Internet telephony services. The IP islands grow evenlarger into a “constellation” of non-connected communities. While suchcommunities can in some cases be linked using the Public SwitchedTelephone Network (PSTN), the benefits of IP telephony—e.g., userpresence, unified communications, user preference, and lower costs maybe sacrificed.

Unlike the PSTN in which users and carriers are easily reachable byanyone on the network, IP telephony is subject to several constraints.First, users are required to have knowledge of whether an IP endpoint isavailable if the full capabilities of IP telephony are to be realized.Also, the knowledge of whether there are multiple IP enabled devices isbeing used by the called party as well as how to reach such devices isneeded. Another constraint is that a single IP “telephone” number is notavailable among the various IP enabled devices; instead, these devicesutilize diverse and complex addresses. Further, determining the identityof the calling party (e.g., caller ID) is an important function.

Based on the foregoing, there is a clear need for an approach thatfacilitates bridging of the IP islands, thereby enabling greaterdeployment of IP telephony. There is also a need for a mechanism toensure compatibility and coordination of IP telephony services amongservice providers. There is a further need for an approach to exploitthe full capabilities of Internet telephony technologies.

SUMMARY OF THE INVENTION

These and other needs are addressed by the present invention, in whichan approach for performing network based packetized voice callprocessing is provided.

According to one aspect of the present invention, a method for providingcommunication services is disclosed. The method includes receiving arequest for establishing a voice call from a source endpoint behind afirst network address translator of a first domain, wherein the requestspecifies a directory number of a destination endpoint within a seconddomain. The method also includes selectively determining a networkaddress for communicating with the destination endpoint based on thedirectory number. Additionally, the method includes determiningexistence of a second network address translator within the seconddomain. Further, the method includes, if the network address can bedetermined, establishing a media path between the source endpoint andthe destination endpoint based on the network address to support thevoice call.

According to another aspect of the present invention, a system forproviding managed communication services is disclosed. The systemincludes a proxy server configured to receive a request for establishinga voice call from a source endpoint behind a first network addresstranslator of a first domain, wherein the request specifies a directorynumber of a destination endpoint within a second domain. The system alsoincludes an address server configured to selectively determine a networkaddress for communicating with the destination endpoint based on thedirectory number. Additionally, the system includes a STUN (SimpleTraversal of UDP (User Datagram Protocol)) server configured to supportdetermination of existence of a second network address translator withinthe second domain. The system further includes a TURN (Traversal UsingRelay NAT (Network Address Translation)) server configured to establish,if the network address can be determined, a media path between thesource endpoint and the destination endpoint based on the networkaddress to support the voice call.

According to another aspect of the present invention, a method ofsupporting communication with a mobile terminal is disclosed. The methodincludes receiving a call establishment request from the mobile terminalduring a cellular call over a cellular network, wherein the mobileterminal provides a first operational mode for communicating over thecellular network and a second operational mode for communicating over awireless access network. The method also includes, in response to thecall establishment request, initiating teardown of the cellular call andestablishment of a voice call over the wireless access network.

According to yet another aspect of the present invention, a system forproviding communication services is disclosed. The system includes meansfor receiving a request for establishing a voice call from a sourceendpoint behind a first network address translator of a first domain,wherein the request specifies a directory number of a destinationendpoint within a second domain. The system also includes means forselectively determining a network address for communicating with thedestination endpoint based on the directory number. Further, the systemincludes means for determining existence of a second network addresstranslator within the second domain; and means for establishing a mediapath between the source endpoint and the destination endpoint based onthe network address to support the voice call, if the network addresscan be determined.

Still other aspects, features, and advantages of the present inventionare readily apparent from the following detailed description, simply byillustrating a number of particular embodiments and implementations,including the best mode contemplated for carrying out the presentinvention. The present invention is also capable of other and differentembodiments, and its several details can be modified in various obviousrespects, all without departing from the spirit and scope of the presentinvention. Accordingly, the drawings and description are to be regardedas illustrative in nature, and not as restrictive.

BRIEF DESCRIPTION OF THE DRAWINGS

The present invention is illustrated by way of example, and not by wayof limitation, in the figures of the accompanying drawings and in whichlike reference numerals refer to similar elements and in which:

FIG. 1 is a functional diagram of a communication system for supportinginterconnectivity of disparate packetized voice networks, according toone embodiment of the present invention;

FIG. 2 is a diagram of a communication system capable of providinginterdomain traversal, according to one embodiment of the presentinvention;

FIG. 3 is a diagram of an exemplary architecture for supporting ENUM(Electronic Number) services in the system of FIG. 1, according to oneembodiment of the present invention;

FIG. 4 is a diagram of an exemplary Session Initiation Protocol(SIP)-to-SIP call flow, according to an embodiment of the presentinvention;

FIG. 5 is a diagram of an exemplary SIP-to-PSTN (Public SwitchedTelephone Network) call flow, according to an embodiment of the presentinvention;

FIG. 6 is a diagram of an architecture utilizing a centralized datastore supporting communication among remote endpoints, according to anembodiment of the present invention;

FIG. 7 is a diagram of a wireless communication system for providingapplication mobility, according to one embodiment of the presentinvention;

FIGS. 8A and 8B are diagrams of exemplary multimodal wireless and wireddevices, according to various embodiments of the present invention;

FIG. 9 is a diagram of a process for authentication and registration ofa multimodal device in a data network, according to one embodiment ofthe present invention;

FIG. 10 is a diagram of a process for establishing a call from amultimodal device to the PSTN, according to one embodiment of thepresent invention;

FIG. 11 is a diagram of a process for establishing a call to amultimodal device from the PSTN, according to one embodiment of thepresent invention;

FIG. 12 is a diagram of a process for cellular-to-IP mode switchingduring a call supported by the PSTN, according to one embodiment of thepresent invention;

FIG. 13 is a diagram of a process for IP-to-cellular mode switchingduring a call supported by the PSTN, according to one embodiment of thepresent invention;

FIG. 14 is a diagram of a process for call establishment by a multimodaldevice operating in cellular mode, according to one embodiment of thepresent invention;

FIG. 15 is a diagram of a process for cellular-to-IP mode switchingmid-call, according to one embodiment of the present invention;

FIG. 16 is a diagram of an Operational Support System (OSS)architecture, according to one embodiment of the present invention;

FIG. 17 is a diagram of a financial system for supporting IPInterconnect service, according to one embodiment of the presentinvention;

FIG. 18 is a diagram of a service assurance infrastructure componentscapable of supporting the Interconnect services, in accordance with anembodiment of the present invention; and

FIG. 19 is a diagram of a computer system that can be used to implementvarious embodiments of the present invention.

DESCRIPTION OF THE PREFERRED EMBODIMENT

An apparatus, method, and software for providing interdomain traversalto support packetized voice transmissions are described. In thefollowing description, for the purposes of explanation, numerousspecific details are set forth in order to provide a thoroughunderstanding of the present invention. It is apparent, however, to oneskilled in the art that the present invention may be practiced withoutthese specific details or with an equivalent arrangement. In otherinstances, well-known structures and devices are shown in block diagramform in order to avoid unnecessarily obscuring the present invention.

Although the various embodiments of the present invention are describedwith respect to the Internet Protocol (IP) based voice sessions, it iscontemplated that these embodiments have applicability to othercommunication protocols.

FIG. 1 is a functional diagram of a communication system for supportinginterconnectivity of disparate packetized voice networks, according toone embodiment of the present invention. An IP interconnect system 100defines an architecture for a “bridging” service (IP interconnect(IP-IC)), for example, to enterprises and service providers for enablingInternet Protocol (IP) telephony communications among these enterprises.The term “IP interconnect” as used herein is a mechanism thatfacilitates IP calling by discovering IP users within a registry 101maintained, for example, by a service provider. The registry is used todetermine how IP calls are routed over the Internet, or where noInternet or alternate IP paths are available, to the PSTN or mobilephones.

It is recognized that development of new Internet technologies hasenabled creation of new communication services. As a result, strictlytraditional communication services over the Public Switched TelephoneNetwork (PSTN) are becoming less attractive economically andfunctionally. Coincident with greater accessibility to the“constellation” of IP endpoints (e.g., VoIP/IM users across enterprise,carrier/ISP and wireless networks), it is recognized that new featuresfor enhancing the IP calling experience can be developed.

The approach, according to an embodiment of the present invention,provides seamless Internet interconnect between enterprise IP islands,and management of the routing and services offered between such islands.Also, the approach supports traffic between IP enabled Private BranchExchange (PBX) systems and endpoints (e.g., Session Initiation Protocol(SIP) clients) over the global Internet and IP islands of other serviceproviders—e.g., cable operators, Internet Service Providers (ISPs),Virtual VoIP service providers, etc.

The IP interconnect service system 100, according to one embodiment ofthe present invention, encompasses the following functional components:a discovery component 103, an identity component 105, a signalingconversion component 107, and a Network Address Translation(NAT)/Firewall traversal component 109. As used herein, the termsNetwork Address Translation or Network Address Translator are usedsynonymously. These functional components (or modules) 103-109 provide acapability for enabling connectivity for multiple IP telephony networks111 a-111 n n behind NAT and/or firewalls 113 a-113 n. The system 100,thus, provides for interdomain traversal across these NAT and/orfirewalls 113 a-113 n.

Firewalls 113 a-113 n provide security for interfacing with anothernetwork (e.g., an untrusted network). It is noted that a private network(e.g., enterprise) having connectivity to external network, such aspublic data network (e.g., the Internet), can be subjected to varioussecurity risks. Firewalls can be implemented as hardware and/or softwareto prevent unauthorized access to the private network. Firewalls monitorincoming and outgoing traffic and filters (or blocks) such trafficaccording to certain rules and policies. A firewall can employ varioustechniques to filter traffic; e.g., packet (or flow) filtering examinespackets to ensure specified requirements are met with respect to thecharacteristics of the packet (or flow). Hence, the process only allowspackets satisfying such requirements to pass. These requirements can bebased on network addresses, ports, or whether the traffic is ingress oregress, etc.

Network Address Translation (NAT) performs translation between privatenetwork addresses and public network addresses; i.e., providing privateaddress to public address binding. This binding can be static ordynamic. In the context of security and firewalls, NAT can hide a set ofhost addresses on the private network behind a pool of public addresses.In this manner, external networks cannot “see” internal addresses, andthereby prevent establishment of connections not originating from theprivate network. The pool can be one or more network addresses, or canbe a range of network addresses (e.g., a set of contiguous networkaddresses). The NAT can also specify a port range to restrict porttranslation. NAT is further detailed in RFC 3022, which is incorporatedherein by reference in its entirety.

As indicated, discovery 103 plays an important part in providing the“bridging” service to IP enabled “islands.” The discovery query can beaccomplished using a DNS (Domain Name Service) query (ENUM) or via a SIPquery (Redirect server). While this discovery mechanism is most usefulbetween islands 111 a-111 n, for the sake of simplicity, this mechanismcan be used for all requests, even those within an island. OnceIP-enabled island discovery is complete, identity 105 is the nextconcern.

A cryptographically secure identity mechanism 105 can prevent, forexample, spam problems confronting email systems. In addition, theidentity service 105 provides a “Caller ID” service on the Internet.

As regard signaling conversion 107, in some cases, IP-enabled islands111 a-111 n are unable to communicate due to different signalingprotocols (e.g., Session Initiation Protocol (SIP) vs. H.323) orprotocol incompatibilities (e.g., stemming from different versions ofSIP). The IP interconnect service provides signaling conversion for allcommon protocols (e.g., SIP and H.323), versions, and dialects. Thisservice can be provided, via a SIP proxy service.

By way of example, the system 100 utilizes IP telephony signaling thatincludes, for example, the H.323 protocol and the Session InitiationProtocol (SIP). The H.323 protocol, which is promulgated by theInternational Telecommunication Union (ITU), specifies a suite ofprotocols for multimedia communication. SIP is a competing standard thathas been developed by the Internet Engineering Task Force (IETF). SIP isa signaling protocol that is based on a client-server model. It shouldbe noted that both the H.323 protocol and SIP are not limited to IPtelephony applications, but have applicability to multimedia services ingeneral. In an embodiment of the present invention, SIP is used tocreate and terminate voice calls over an IP network. However, it isunderstood that one of ordinary skill in the art would realize that theInternational Telecommunications Union (ITU) H.323 protocol suite andsimilar protocols can be utilized in lieu of SIP.

The IP interconnect service enables the creation of innovative IP-basedservices that add value to the user, beyond Internet calling, bydefining powerful call preference capabilities. Within the service,Voice over IP (VoIP), Instant Messaging (IM), conferencing,collaboration, and other IP communication services are supported.

FIG. 2 is a diagram of a communication system capable of providinginterdomain traversal, according to one embodiment of the presentinvention. The communication system 200 supplies IP interconnectservices, according to the functional architecture of the system ofFIG. 1. In an exemplary embodiment, the system 200 provides ENUM serviceand NAT/Firewall traversal, via an ENUM server 201, a STUN (SimpleTraversal of UDP (User Datagram Protocol)) server 203 and a TURN(Traversal Using Relay NAT) server 205. Where NAT and firewall traversalis required, the IP interconnect service provides both endpointinitiated services (e.g., STUN and TURN servers 203, 205) and networkinitiated services (e.g., ALG (Algorithm) and proxy services).

According to one embodiment of the present invention, the serviceprovider system 200 offers an open managed service for the interdomaintraversal. This approach contrasts with the traditional traversal, whichis controlled by supernoding (other users) or session border controllersin one domain or the other. Interdomain traversal supports establishinga peer-to-peer communication session between two distinct virtuallocations (or domains 207, 209) separated by firewalls 207 a, 209 aand/or NATs 207 b, 209 b. Procession of call flows managed serviceenables the interdomain traversal: ENUM Service. Interdomain traversalinvolves communicating between a device in one administrative domain 207and another device in a different administrative domain 209.

In an exemplary embodiment, a SIP proxy server (e.g., servers 207 e and209 e) maintains registration for all users in its domain, as well asdirectory numbers (i.e., telephone numbers) for them. Upon receiving arequest for the directory number, if the SIP proxy server determinesthat number does not correspond to one of registered users, the SIPproxy server queries the ENUM server 201 to obtain the requested number.

In an exemplary embodiment, the system 200 supports customization ofcomponents and processes to enable procession of call flows managedservice; these components include the client (e.g., 207 c and 207 d),SIP proxy server 207 e, and TURN server 205. The SIP proxy server 207 emaintains the user ID's along with their assigned telephone numbers. Aregistry (not shown) contains identifiers (including aliases) andassociated telephone numbers. The SIP proxy server 207 e can beconfigured with routing rules. For example, the SIP proxy server 207 emay require looking through the list of registry first before queryingthe ENUM server 201. If found in the ENUM server 201, the URIcorresponding to the telephone number is obtained from the server 201.The URI can be utilized for the INVITE onto the appropriate SIP proxyserver (e.g., 209 e) for that domain (e.g., 209). The registry ofaliases and associated telephone numbers can be maintained locally tominimize querying the ENUM server 201. Essentially, the contactinformation from the ENUM server 201 is cached for subsequent use,thereby minimizing network traffic and processor loads on the ENUMserver 201. With respect to the client 207 c, 207 d, configuration ismade so that the client 207 c, 207 d knows the location of the TURNserver 205. No code modification is required; by default, the client 207c, 207 d can be configured to try to communicate with the SIP proxyserver 207 e or session border controller.

Providing the TURN server 205 as a managed service involves setting upcredentials for users. The SIP proxy server 207 e can maintaincredentials for users and be managed by an enterprise. In managedservice network 200 (i.e., “cloud”) of the service provider, credentialpairs are utilized, as enterprise users may not want SIP Usercredentials to be managed by the service provider.

The Traversal Using Relay NAT (TURN) protocol permits an element behinda NAT and/or firewall to receive incoming data over Transmission ControlProtocol (TCP) or User Datagram Protocol (UDP) connections. That is, thenetwork element within the private network can be on the receiving end,rather than the sending end, of a connection that is requested by thehost.

STUN is a lightweight protocol that allows applications to discover thepresence and types of Network Address Translators and firewalls betweenthem and the public Internet. This protocol also provides the abilityfor applications to determine the public IP addresses allocated to themby the NAT. STUN allows a wide variety of applications to work throughexisting NAT infrastructure.

According to various embodiments of the present invention, the IPinterconnect service employs standards-based ENUM and SIP services. Thefunctional structure of the IP interconnect is compatible with, forexample, the Internet DNS and infrastructure domain e164.arpa so thatfuture number records migration can be performed seamlessly coincidentwith public ENUM deployment.

ENUM provides translation of telephone numbers (e.g., E.164) intoUniform Resource Identifiers (URIs), thereby communication with an IPendpoint. It is noted that ENUM is “protocol agnostic” because it isapplication agnostic, and thus, operates with either H.323 or SIP.

ENUM is a protocol that resolves fully qualified telephone numbers(e.g., E.164) to fully qualified domain name addresses using a DomainName System (DNS)-based architecture. The protocol, as defined in RFC2916, uses the DNS for storage of E.164 numbers and supports servicesassociated with an E.164 number. E.164 refers to the internationaltelephone numbering plan administered by the InternationalTelecommunication Union (ITU). E.164 specifies the format, structure,and administrative hierarchy of telephone numbers. A fully qualifiedE.164 number is designated by a country code, an area or city code, anda phone number.

The translation of a telephone number into an Internet address proceedsas follows. A fully qualified number has the form: “+1-234-567-8910.”First, non-numerical characters are removed: 12345678910. Next, theorder of these digits are reversed: 01987654321. Thereafter, decimalpoints are introduced between the digits, resulting in“0.1.9.8.7.6.5.4.3.2.1,” and the domain “e164.arpa” is appended. Thisyields “0.1.9.8.7.6.5.4.3.2.1.e164.arpa.” The .arpa domain has beendesignated for Internet infrastructure purposes. Based on this address,the ENUM protocol issues a DNS query, and retrieves the appropriateNAPTR (Naming Authority Pointer) Resource records, which containinformation about what resources, services, and applications areassociated with a specific phone number. These services are determinedby the subscriber.

By way of example, the system 200 ensures communication betweendifferent IP telephony networks, which reside in differentadministrative domains 207 and 209, over a public data network 211, suchas the global Internet. The network within domain 207 includes afirewall 207 a for interfacing the public data network 211. Behind thefirewall 207 a is a NAT 207 b that serves a variety of endpoints capableof supporting IP telephony—e.g., a web phone 207 c, and a so-called“soft” phone 207 d. The network also utilizes a proxy server 207 e forsupporting packetized voice calls, which in this example is compatiblewith SIP.

As regard the network 209, a firewall 209 a resides between the network209 and the pubic data network 211. A NAT 209 b serves a soft phone 209c and one or more SIP phones 209 d. The network 209 e also includes aSIP proxy server 290 e.

As shown, the Internet 211 communicates with a circuit switchedtelephone network 213, such as the PSTN, through a gateway 215. Underthis scenario, the PSTN 213 supports cellular capable devices 217 (e.g.,cellular phones) as well as POTS (Plain Old Telephone Service) phones219.

FIG. 3 is a diagram of an exemplary architecture for supporting ENUMservices in the system of FIG. 1, according to one embodiment of thepresent invention. In one scenario, the system 200 of FIG. 2 includes anENUM system 300 employing various ENUM components, such as an ENUM DNSRoot server 301, an ENUM DNS Tier 2 server 303, and ENUM Redirect server305. The system 300 also includes a Proxy/Authentication server 307, anAAA server 309, a Certificate Store/Authority component 311, andSignaling Conversion gateways 313 and 315 (i.e., H.323-to-SIP Gateway313 and SIP-to-SIP gateway 315). Additionally, a SIP Network-based NATTraversal is provided. Further, the system 300 utilizes the STUN server203, a Media Relay server 316, and a Service Oriented ArchitectureInformation Technology (SOA IT) 317.

The Media Relay Server 316 and two user agents (UAs) in either domainpass each other information about their environment. Such informationcan include external firewalls, internal IP addresses, and supportinformation.

The ENUM DNS Root server 301 provides a combined Tier 0/Tier 1 ENUM rootfunctionality. Because country codes may not be generally available, theservice provider can host its own ENUM tree; this can be structured in asimilar way to the e164.arpa tree. According to one embodiment of thepresent invention, this root server 301 supplies DDOS (DistributedDenial of Service) protection. According to an embodiment of the presentinvention, the ENUM DNS Root server provides ENUM services according toRFC 3761 and RFC 2916, which are both incorporated herein by referencein their entireties.

The ENUM DNS Tier 2 server 303 is the DNS functionality that containsactual DNS NAPTR records—e.g., one per telephone number. It is notedthat only E.164 (global) telephone numbers are used—no private numbers.These records are created and backed up by an administrative system thatwill tie into the order entry and billing systems (as later describedwith respect to FIG. 17). It is assumed that entries are authorized andvalidated using various mechanisms, which can include knownauthorization and validation standards. The NAPTR records can be queriedby any IP-enabled endpoint or island on the Internet, regardless ofwhether they are an IP interconnect customer or not. In this way, thediscovery service mimics that of the public ENUM.

The ENUM DNS Tier 2 server 303, in an exemplary embodiment, utilizesexisting DNS server farms to implement the ENUM Tier 2 functionality. Aprovisioning system (such as that of FIG. 17) can collect the telephonenumber to URI mapping information from the IP interconnect customers andautomatically generate the NAPTR records. As Public ENUM is deployed,the service provider can become a Tier 2 provider in each country code.The provisioning interface can then be adapted to interface with eachTier 1 function. According to an embodiment of the present invention,the ENUM DNS Tier 2 server provides ENUM services according to RFC 3761as well as RFCs 3762 and 3764 (which are incorporated herein byreference in their entireties).

The ENUM SIP redirect server 305 behaves as a SIP redirect server byaccepting SIP requests and responding with a 3xx class response, forexample. According one embodiment of the present invention, thisredirect server 305 has a built-in ENUM resolver, and queries the ENUMTier 2 Server using DNS. That is, the server 305 can perform ENUMqueries for IP-enabled endpoints or islands that do not have an ENUMresolver; the resolver takes a telephone number, performs a DNS query,and returns a set of Uniform Resource Identifiers (URIs). For instance,the ENUM Redirect server 305 accepts a SIP request (such as INVITE,SUBSCRIBE, or even other methods such as OPTIONS), performs an ENUMquery on the telephone number in the Request-URI, and returns a redirectresponse (302 Moved Temporarily or 300 Multiple Choices) containingContact header fields with each resolved URI.

For the purposes of explanation, the Request-URI can be a tel URI(tel:+13145551234) or a SIP URI with a telephone number in the user part(sip:+458320923@mci.com;user=phone). The telephone number, in anexemplary embodiment, is in E.164 (global) format. If an endpoint is notable to generate requests in this format, the SIP-to-SIP Gateway servicecan be used to generate this format. The time-to-live (TTL) informationin the ENUM record are translated into an expires parameter for eachURI. It is noted that non-SIP URIs may be returned. The resulting set ofURIs are mapped into SIP Contact header fields and returned.

If a single URI is returned, it can be done so in a 302 MovedTemporarily response. If multiple URIs are to be returned, a 300Multiple Choices response is returned. Other SIP elements such as theProxy/Authentication Server 307, H.323-to-SIP Gateway 313, andSIP-to-SIP Gateway 315 all interact with the ENUM Redirect server 305using standard SIP messages. It is noted that the ENUM Redirect server305 does not perform any resolution on the URIs from the ENUM query—theyare passed unchanged in the redirect response. If the ENUM query failsto return any URIs, the ENUM Redirect server 305 returns a single telURI representing the telephone number in the Request-URI. If theRequest-URI does not contain a valid E.164 telephone number, the serverreturns a 404 Not Found response.

The Proxy/Authentication Server 307 is the SIP edge of the IPinterconnect service. The Proxy/Authentication Server 307 has two keyfunctions, authentication and proxying requests. The authenticationfunction can be provided on behalf of other elements in thearchitecture, such as the ENUM Redirect server 305.

The authentication method is determined by the type of security on thelink from the service provider to IP interconnect. If the SIP requestarrives over a Transport Layer Security (TLS) connection, thecertificate provided may be use for authentication. The certificate maybe one issued by the Certificate Authority (CA)/Store or it may be oneissued by another CA. If the SIP request comes in over a Virtual PrivateNetwork (VPN) or IPSec (IP Security), then the use of the private keyprovides authentication. Otherwise, the request receives a SIP Digestchallenge in the form of a 407 Proxy Authentication Required responsecontaining a one time nonce.

The Transport Layer Security (TLS) Protocol provides privacy and dataintegrity between two applications, and has two layers: the TLS RecordProtocol and the TLS Handshake Protocol. The TLS Record Protocol resideson top of a reliable transport protocol, such as TCP. The TLS RecordProtocol provides connection security. Symmetric cryptography is usedfor data encryption (DES, RC4, etc.); the keys are generated uniquelyfor each connection and are based on a secret negotiated by anotherprotocol (such as the TLS Handshake Protocol). The Record Protocol canalso be used without encryption. The message transport includes amessage integrity check using a keyed Medium Access Control (MAC),wherein secure hash functions (e.g., SHA, MD5, etc.) are used for MACcomputations. The TLS Record Protocol provides encapsulation of varioushigher level protocols, such as the TLS Handshake Protocol. The TLSHandshake Protocol allows the server and client to authenticate eachother and to negotiate an encryption algorithm and cryptographic keysbefore the application protocol transmits or receives its first byte ofdata.

The TLS protocol is detailed in RFCs 2246 and 3546 (which areincorporated herein by reference in their entireties); this securityprotocol is formerly known as the Secure Sockets Layer (SSL).

The Proxy/Authentication Server 307 compares the re-sent request withthe MD5 hash of the shared secret to the shared secret retrieved fromthe AAA server 309. A match provides authentication. An authorizationfailure will result in a 403 Forbidden response being sent.

Once authentication has succeeded, the Proxy/Authentication Server 307can provide identity services (as described in FIG. 1). Before anyidentity services are performed, the From header URI is compared to alist of valid identities for the authenticated party. It is noted thatthis scope will typically be restricted to the domains of record (hostpart, not user part) and telephone numbers in tel URIs. If the Fromidentity is valid, identity services may be performed. If it is notvalid, a 403 Invalid From Identity response is returned and no furtherservices are rendered.

It is noted that the presence of a Privacy header field in the requestmay override the normal identity assertion rules. However, the IPinterconnect service does not provide complete IP privacy by itself,although using TURN it may be possible for an endpoint to establish atruly private IP session.

According to one embodiment of the present invention, the followingidentity options are provided: Authenticated Identity Body (AIB),P-Asserted-Identity, and Identity. The particular method that isrequested is based on the authenticated user's service profile. Inaddition, a user's profile will indicate the default server option toproxy or redirect. Alternatively, SIP caller preferences can be used toindicate which mode of operation is desired on a request by requestbasis.

For the AIB method, the Proxy/Authentication Server 307 generates anAuthenticated Identity Body (AIB) and returns it in a 302 MovedTemporarily response. The AIB is signed by the Proxy/AuthenticationServer 307 using the IP interconnect private key. The resulting requestis then retried by the user with the AIB included as a message body. TheAIB method is used in a redirect mode.

For the P-Asserted-Identity method, the Proxy/Authentication Server 307generates the P-Asserted-Identity header field, possibly using theP-Preferred-Identity header field if multiple identities are valid. TheP-Asserted-Identity method is used in proxy mode. An additionalrequirement on P-Asserted-Identity is the use of an integrity protectedSIP connection from the Proxy/Authentication Server 307 and the next hop(effectively this means TLS transport or the use of VPN or IPSectunnel). If integrity protection is not available, noP-Asserted-Identity service can be provided.

For the Identity method, the Proxy/Authentication Server 307 generatesan Identity header field and either returns it in a redirect or proxiesthe request. The Identity method can be used in either proxy or redirectmode. In proxy mode, the Proxy/Authentication Server 307 performs DNSresolution on the Request-URI according to normal SIP DNS rules andprepares to proxy the request.

The Proxy/Authentication Server 307 has SIP interfaces to the ENUMRedirect server 305, the H.323-to-SIP gateway 313, and the SIP-to-SIPgateway 315. Authentication can be performed, according to an exemplaryembodiment, using normal SIP mechanisms, such as SIP Digest challenge,certificate validation, or symmetric key encryption (e.g., IPSec orVPN). Credentials are verified in a AAA database using the RADIUSprotocol.

Additionally, the Proxy/Authentication Server 307 can serve as one ormore SIP servers 317 and 319 (or “soft switches”).

The Authentication, Authorization, and Accounting (AAA) Server 309provides various service specific information such as credentials,preferences, and service options. The AAA server 309 stores the sharedsecrets (usernames/passwords) of IP interconnect customers. This server309 is accessed by other elements using RADIUS—e.g., theProxy/Authentication Server 307, SIP to H.323 Gateway, SIP-to-SIPgateway 315, and TURN Servers. SIP AAA functions are further detailed inRFC 3702, which is incorporated herein by reference in its entirety.

The Certificate Store/Authority server 311 hosts and allocatescertificates to IP-enabled endpoints or islands. The certificates can bestored locally on the respective islands or can be stored in thenetwork. The Certificate Authority (CA) Store 311 provides certificatecreation, management, revocation, storage and distribution. Thecertificates can be either self-signed certificates (suitable forindividual SIP endpoints to use for Secure/Multipurpose Internet MailExtensions (S/MIME) or SRTP (Secure Real-time Transport Protocol)) orcertificates issued by the IP interconnect CA. By way of example, thecertificates can be fetched using TLS, SIP and HyperText TransferProtocol (HTTP)-based mechanisms. The Certificate Authorityfunctionality provides limited SIP identity assertions, and thus,provides a more cost-effective approach than conventional Verisign-typee-commerce certificates.

In addition, the Proxy/Authentication Server 307 uses the CertificateAuthority/Store to retrieve and verify certificates of customers.

The H.323-to-SIP gateway 313, in this example, provides conversionbetween H.323 and SIP. According to one embodiment of the presentinvention, This gateway 313 can serve an IP PBX 321. To the SIP network,the gateway 313 appears as a SIP User Agent, while appearing as a H.323Gatekeeper to a H.323 network. Normal H.323 authentication mechanismscan be used.

Under the scenario of FIG. 3, a SIP-to-SIP gateway 315 for convertingincompatible SIP dialects to, for example, the standard RFC 3261 SIP.Some typical “broken” SIP issues include incorrect use of To/From tags,malformed header fields and bodies, nonstandard methods, nonstandardDTMF transport methods, multipart Multipurpose Internet Mail Extensions(MIME) handling issues (e.g., SIP-T (Session Initiation Protocol forTelephones)), proprietary authentication schemes, transport protocolincompatibilities, improper Record-Route and proxy routing behavior, andIPv6 to IPv4 mapping.

The SIP-to-SIP gateway 315 acts as transparently as possible, whenserving IP PBX 323, for example. The SIP-to-SIP gateway 315 alsoprovides the authentication function, and support some additionalauthentication schemes. According to an embodiment of the presentinvention, credentials are verified in a AAA database using the RADIUSprotocol. This protocol can be embedded in various network elements:routers, modem servers, switches, etc. RADIUS facilitates centralizeduser administration, which is important in large networks havingsignificant number of users. Additionally, these users are continuallybeing added and deleted (resulting in constant flux of authenticationinformation). RADIUS is described in Internet Engineering Task Force(IETF) Request For Comment (RFC) 2865 entitled “Remote AuthenticationDial In User Service (RADIUS)” (June 2000), which is incorporated hereinby reference in its entirety.

The SIP Network-based NAT Traversal function performs the necessarysignaling to support network based NAT traversal by invoking a mediarelay function (e.g., TURN or RTP proxy) for sessions that wouldotherwise fail. According to an embodiment of the present invention,only islands provisioned for this service can utilize this function.Network-based NAT traversal is provided when the island does not managethis function internally. When a media relay is required, the SIP-to-SIPgateway 315 invokes one from the Media Relay function, and modify theSIP signaling messages appropriately. In addition to TURN, otherprotocols can be used between the SIP-to-SIP gateway 315 and the MediaRelay 316.

It is noted that this SIP Network-based NAT Traversal function istransparent to islands using STUN and TURN—this appears as if no NAT ispresent, and hence no action is taken. The NAT traversal functionalitycan be provisioned for a given island rather than dynamically detected.This is because the dynamic detection of NATs requires registration datawhich is generally not available from islands.

The Simple Traversal of UDP through NAT (STUN) Server 203 providesendpoint-based NAT discovery and characterization. A STUN-enabledendpoint can traverse most NAT types without relying on network-baseddetection and fixing. An endpoint can determine the type of NAT (e.g.,full cone, restricted cone, or symmetric) and discover and maintainbindings between private and public IP addresses. For an endpoint, thecombination of STUN and TURN usage, as described in the ICE(Interactivity Communication Establishment) protocol, provides completeendpoint-based NAT traversal.

It is noted that the STUN server 203 does not authenticate users,largely because the resources used are trivial as it is essentially justa type of “ping” server. As a result, no AAA or provisioning tie in isnecessary. STUN server discovery can be provided using DNS SRV lookupson the domain used by the IP interconnect service. The STUN functionsare further detailed in RFC 3489, which is incorporated herein byreference in its entirety.

The Media Relay function provides the relay functionality needed incertain NAT and firewall traversal scenarios. This function is providedusing both TURN (Traversal Using Relay NAT) Server 205 (forendpoint-enabled traversal) and RTP proxies (for network-based relay).Authentication is performed using SIP Digest credentials and accessedusing RADIUS from the AAA server 309. In an exemplary embodiment, theMedia Relay function provides RTP and Real-Time Control Protocol (RTCP)relay functionality for NAT and firewall traversal.

According to one embodiment of the present invention, the Media Relayfunction is decentralized and distributed throughout the serviceprovider's IP backbone. In addition, some optimal Media Relay selectionalgorithms can be used. In the alternative, centrally deployed mediarelays can be utilized if a distributed architecture cannot be achieved.The architecture supports both network invoked and endpoint invokedmedia relay functionality. As such, a standards-based protocol, such asTURN, is used. Media Relays are a significant network resource; as such,they must authenticate and account for usage. Because the TURN functionsupports reuse of existing SIP Digest credentials, the TURN servers areable to access the AAA Servers (e.g., server 309).

The SOA IT Server 317 provides the “back office” functions necessary toprovide the Interconnect service. That is, the SOA IT has componentsthat provide the Operational Support System (OSS) functions needed torun and support the IP interconnect product offering as arevenue-generating business. According to one embodiment of the presentinvention, the SOA IT components include both customer-facing systems(e.g., enabling customer self-service), and back-office systems. The SOAIT components largely concentrate on the so-called F-A-B broadfunctional areas: Fulfillment, Assurance and Billing—as well as ensuringthat such functions are compliant with regulatory reportingrequirements. Such functions are more fully described with respect toFIG. 17.

The described IP interconnect services involve the interaction of SIP,STUN and TURN protocols to support IP telephony. This interaction isexplained in the call flows of FIGS. 4 and 5, in the context of FIG. 2.

FIG. 4 is a diagram of an exemplary Session Initiation Protocol(SIP)-to-SIP call flow, according to an embodiment of the presentinvention. For the purposes of illustration, the source (or originating)endpoint is the soft phone 207 d and has an identifier,bob@voiptheworld.net. The destination (or terminating) endpoint is thesoft phone 209 c with user, alice@ipislands.com. In step 401, theendpoint 207 d establishes communication with the STUN server 203 byissuing a binding request. This communication is established using astandard TCP handshake and authentication process (step 403). Next, theendpoint 207 d sends a register signal, e.g., using SIP (REGISTER/200OK), to the SIP proxy server 207 e using a connection through the TURNserver 205 (step 405). The SIP proxy server 207 e responds, as in step407, with a 200 OK message to the endpoint 207 d.

In step 409, the endpoint 207 d submits an INVITE message to the SIPproxy server 207 e, which replies with a 100 Trying message (step 411).

At this point, the proxy server 207 e determines that the URI of thedestination endpoint 209 d needs to be determined. Accordingly, the SIPproxy server 207 e submits a DNS query to the ENUM server 201, whichresponds with the appropriate NAPTR record (steps 413 and 415).

Next, the SIP proxy server 207 e sends the INVITE message to the SIPproxy server 209 e of the destination network (step 417). The SIP proxyserver 209 e forwards the INVITE message to the destination endpoint 209d, per step 419.

The endpoint 209 d then sends a 180 Ringing message, as in step 421, tothe SIP proxy server 209 e, which relays the message to the SIP proxyserver 207 e (step 423). Thereafter, the Ringing message is transmitted,per step 425, to the source endpoint 207 d.

In step 427, the endpoint 209 d generates a 200 OK message, forwardingthe message to the SIP proxy server 209 e. In step 429, this 200 OKmessage is relayed by the SIP proxy server 209 e to the other SIP proxyserver 207 e. Thereafter, the 200 OK message is forwarded by the SIPproxy server 207 e to the source endpoint 207 d, as in step 431. Theendpoint 207 d acknowledges the SIP proxy server 207 e with an ACKmessage (step 431). The SIP proxy server 207 e sends the ACK message tothe destination endpoint 209 d through the SIP proxy server 209 e (steps435 and 437). In step 439, the endpoints 207 d and 209 d now canexchange media via the TURN server 205.

FIG. 5 is a diagram of an exemplary SIP-to-PSTN (Public SwitchedTelephone Network) call flow, according to an embodiment of the presentinvention. Under this scenario as with the SIP-to-SIP call flow,communication is performed via the TURN server 205. The endpoint 207 destablishes communication with the STUN server 203 with a bindingrequest, per step 501. A standard TCP handshake and authenticationprocess is executed, per step 503, between the endpoint 207 d and theSTUN server 203. The endpoint 207 d transmits a register signal to theSIP proxy server 207 e (step 505). The SIP proxy server 207 e sends a200 OK message to the endpoint 207 d in response to the Register signal,per step 507.

In step 509, the endpoint 207 d sends an INVITE message to the SIP proxyserver 207 e. The server 207 e then replies with a 100 Trying message(step 511).

Per step 513, the proxy server 207 e sends a DNS query to the ENUMserver 201. In this example, the ENUM server 201 cannot find thecorresponding URI, and indicates so to the SIP proxy server 207 e, perstep 515. Accordingly, the SIP proxy server 207 e sends an INVITEmessage to the media gateway 215 (step 517); the INVITE messagespecifies the telephone number. The media gateway 215, as in step 519,replies with a 180 Ringing message. The SIP proxy server 207 e forwardsthe 180 Ringing message to the endpoint 207 d, per step 521.

In step 523, the media gateway 215 also sends a 200 OK message to theSIP proxy server 207 e. This message is then forwarded to the endpoint207 d (step 525) by the SIP proxy server 207 e.

The endpoint 207 d responds with an ACK message to the SIP proxy server207 e, which sends the message to the media gateway 215 (steps 527 and529). In step 531, a call is established between the source endpoint 207d and the PSTN via the media gateway 215.

FIG. 6 is a diagram of an architecture utilizing a centralized datastore supporting communication among remote endpoints, according to anembodiment of the present invention. A communication system 600 includesa service provider network 601 deploying components to support theInterconnect services, as described above. Notably, the network 601utilizes a data store 603 (or registry) to manage communication amongthe endpoints 605, 607 and 609. These endpoints 605, 607 and 609, forexample, can be associated with a single enterprise, organization orentity, in which the endpoint 605 can correspond to an office location,the endpoint 607 with the home, and the endpoint 609 with a temporary,mobile location such as a hotel.

The data store 603 stores user information as well as information on howpacketized voice calls are to be routed over a public data network suchas the Internet; further, this registry 603 can specify alternate paths,including circuit-switched paths, cellular paths, or IP media paths;such routing information can take many forms, including networkaddresses, protocol port information, etc. Additionally, the data store603 permits the service provider to store and manage billing and ratinginformation for calls placed by users. Further, the service provider canmaintain the necessary information to authorize communication betweenthe end points involving different network elements.

The network 601 includes a SIP proxy server 611 for interfacing thevarious endpoints 605, 607 and 609. The SIP proxy server 611 interactswith a TURN server 613, a STUN server 615 and an ENUM server 617 asdetailed early for supporting packetized voice calls with other datanetworks as well as circuit-switched telephone systems.

In addition, the system 601 utilizes a gateway 619 to provideconnectivity to other systems (e.g., data network or circuit switchedtelephone network).

It is contemplated that the above architecture can be deployed in avariety of terrestrial and radio communication systems to offer theInterconnect services, which can be complementary or supplementary toother communication services. For example, a wireless communicationsystem can implement such services, as explained below.

FIG. 7 is a diagram of a wireless communication system for providingapplication mobility, according to one embodiment of the presentinvention. In accordance with an embodiment of the present invention,the Interconnect services can be deployed in a wireless and wired system700 for providing SIP-based mobile IP communication services. As shown,one or more multimodal mobile devices 701 can communicate using variouswireless technologies—e.g., Wi-Fi™/WiMax, 802.11 or cellular. Under thisscenario, the multimodal device 701 can interface with either a mobiletelephony (e.g., cellular) network 703 or a wireless data network 705.Each of these networks 703 and 705 communicates with a public datanetwork 707, such as the Internet. A service provider network 709 alsohas connectivity to the Internet 707, which communicates with a PublicSwitched Telephone Network (PSTN) 711.

The approach, in an exemplary embodiment, adheres to the followingassumptions. First, the IP side controls all fixed and mobile services.Also, it is assumed that calls are established over a myriad ofnetworks: the Internet 707, 2G/3G mobile networks (3GPP and 3GPP2) 703,Time Division Multiplexing (TDM) networks 714, such as the PSTN and PBXsand ISDN (Integrated Digital Services Network), 4G (4^(th) Generation)Wi-Fi™ and WiMax wireless networks, and IP PBXs and other IP systems,such as H.323. Communication services are enabled or deployed on the IPside and can be based, for instance, on SIP and its associatedapplication layer protocols, such as developed in the SIMPLE, SIPPING,IPTEL, XCON and ENUM working groups of the Internet Engineering TaskForce (IETF). The system 700, for example, includes SIP telephony and IMdevices that are endpoints on the Internet 707. Gateways to 2G/3G mobilephone networks are also endpoints on the Internet 707. Further, SIP-PSTNand SIP-PBX are endpoints on the Internet 707. The above approach iscompatible with the end-to-end applications control architecture of theInternet 707—e.g., IETF documents RFC 3665 and RFC 3666 show exemplarySIP call flow implementations for PBX/Centrex-like telephony andSIP-PSTN, respectively; these documents are incorporated herein byreference in their entireties.

The wireless network 705 (which is a “Visited” network with respect tothe service provider network 709) includes an access point 713 (e.g.,Ethernet switch) as well as an AAA server 715. Likewise, the serviceprovider network 709 includes an AAA server 717. In addition, thenetwork 709 provides a STUN/TURN server 719; these two functions canalso be implemented as separate components, as evident from the previousdiscussion of STUN and TURN functionalities. Further, the serviceprovider network 709 includes a SIP proxy server 721.

The mobile telephony network (e.g., cellular network) 703 includes amobile switch 723 for processing communication sessions from themultimodal mobile station 701 to the PSTN 711 or the Internet 707through a mobile gateway 725. Similarly, a gateway 727 is employed toconnect from the PSTN 711 to the Internet 707; in this manner, thestation 729 within the PSTN 711 can be reached by calls placed over theInternet 707.

Depending on the capabilities supported by the wireless or wired accessnetwork, rich services, such as presence, events, instant messaging,voice telephony, video, games and entertainment services can besupported by the service provider network 709.

It is recognized that modern communication technologies have affordedusers with a multitude of alternatives for communicating. Given thesemany possibilities, a user is unsure, at times, of the most appropriate,expedient way to communicate with another user—given each party'spreferences of when and how to be reached. Users of traditionaltelephone services and Private Branch Exchanges (PBXs) in theenterprise, as well as mobile and Internet communications have atpresent separate devices, identities and subscriptions for eachcommunication service. These users can possess, for example, a homephone, (often with separate local and long distance service), a PBXphone at work, a mobile phone and such as a Personal Digital Assistant(PDA) that may also have mobile phone network and Wireless Local AreaNetwork (WLAN) access to the Internet 707 or to the enterprise PBX.

Additionally, users of Instant Messaging (IM) may also have severalaccounts that can be used with a PC or laptop computer. Likewise, usersof e-mail and mobile Short Messaging Systems (SMS) may also usededicated devices and networks for each particular system, though somebridging between e-mail systems and separately between SMS and IM issometimes possible. Separate subscriptions and mobile devices for accessto these services is still required.

According to one embodiment of the present invention, seamlesscommunications (using presence, SIP events, text, voice, videocommunications and file sharing) is enabled in conjunction with a singleidentity or a suite of similar identifiers. That is, the multimodaldevice 701 enables a user to have a single identity and a single servicesubscription on all mobile and fixed networks, whereby the device 701can operate in dual modes to communicate using any wireless or wirednetwork. One single identity can take the form of a phone number and/ora URI (same or similar to the e-mail address) for all fixed and mobilenetworks and for all types of communications. The phone number and/orURI can be the only entry in the address book, by which the called partycan be both reached and identified. A single identity is provided forthe caller for access to all wired and wireless networks. Also, a singlesubscription can be utilized for all types of networks and devices.Further, NAT and firewall traversal is transparent to the user. Securecommunications can be achieved based on network asserted user identityand encryption on demand.

The mobile device 701 can interwork with PBXs (not shown) or can providePBX-like services. Calls and conferences can be maintained whileswitching between the wireless networks 705 (e.g., 2G/3G (2^(nd)Generation/3^(rd) Generation) mobile phone networks 703, Wi-Fi™/WiMaxwireless broadband) and a wired PSTN 711 (or PBX network).

The Presence, Events, and IM Gateway 319 provides gateway services fromSIP to and from other protocols to enable seamless and interoperablepresence, events, and instant messaging (IM). Presence, events andinstant messaging (IM) have evolved as core new communication serviceson the Internet and in private IP networks with hundreds of millionusers worldwide. Leading edge mobile phone services, such aspush-to-talk are based on presence, events and IM. It is no coincidencethat telephony has become an adjunct to popular IM services, wheremaking a phone call is just another option to choose from various othercommunication modes. IP-IP voice calls are also enabled, without the useof telephone network or dependency on phone numbers.

In both wired and wireless networks, Graphical User Interfaces (GUIs)with the presence of “buddies” can be more useful than displaying phonenumbers. That is, the clicking on presence icons is perceived as moreuseful than using the dial pad. The dial pad remains an option whenconnecting to traditional TDM networks using phone numbers only.

The IM infrastructure is completely separate from other forms ofcommunications, such as voice, video, conferencing, etc. Conventionally,IM services are proprietary and require gateways for at least somedegree of basic communications between disparate systems.

The adoption of the SIP IM Protocols Leveraging Extensions (SIMPLE) bythe mobile industry in the 3G IMS (Third Generation IP MultimediaService) platform as well as by large technology vendors is due to thedesire to have a single SIP based communication infrastructure for allIP communication services.

Gateways between legacy IM protocols can be provided as a fully meshedarchitecture, where the number of gateways increases by the square ofthe number of protocols. However, migration to a common IM core based onSIMPLE standards is a more effective approach and provides gatewaysbetween legacy IM systems and SIMPLE. Under such a scenario, theincrease in gateways is only linear with the number of IM protocolsutilized.

The IM architecture, according to an embodiment of the presentinvention, is based on the SIMPLE standards. The presence event packagedescribes the usage of the Session Initiation Protocol (SIP) forsubscriptions and notifications of presence. Presence is defined as thewillingness and ability of a user to communicate with other users on thenetwork. The presence event package and associated notifications aremore detailed, respectively in “A Presence Event Package for the SessionInitiation Protocol (SIP)” by J. Rosenberg, Internet Draft, IETF work inprogress, January 2003; and “Functional Description of EventNotification Filtering” by H. Khartabil et al., Internet Draft, IETFwork in progress, August 2004 (both of which are incorporated herein byreference in their entireties). Traditionally, presence has been limitedto “on-line” and “off-line” indicators; the notion of presence here isbroader. Subscriptions and notifications of presence are supported bydefining an event package within the general SIP event notificationframework.

The filtering of event notifications refers to the operations asubscriber performs in order to define filtering rules associated withevent notification information. The handling of responses tosubscriptions carrying filtering rules and the handling of notificationswith filtering rules applied to them is defined. The definition alsodescribes how the notifier behaves when receiving such filtering rulesand how a notification is constructed.

The watcher information date format defines template-package for the SIPevent framework. Watcher information refers to the set of userssubscribed to a particular resource within a particular event package.Watcher information changes dynamically as users subscribe, unsubscribe,are approved, or are rejected. A user can subscribe to this information,and therefore learn about changes to it. This event package is atemplate-package because it can be applied to any event package,including itself. Watcher functions are further detailed in “A WatcherInformation Event Template-Package for SIP” by J. Rosenberg, InternetDraft, IETF work in progress, January 2003 (which is incorporated hereinby reference in its entirety).

In particular, the Presence Information Data Format (PIDF) defines abasic format for representing presence information for presentity. Thatformat defines a textual note, an indication of availability (open orclosed) and a URI for communication. However, it is frequently useful toconvey additional information about a user that needs to be interpretedby an automaton, and is therefore not appropriate for placement in thenote element of the PIDF document. Generally, the extensions have beenchosen to provide features common in existing presence systems at thetime of writing, in addition to elements that could readily be derivedautomatically from existing sources of presence, such as calendaringsystems, or sources describing the user's current physical environment.

The Presence Information Data Format (PIDF) defines a basic XML formatfor presenting presence information for presentity. The ExtensibleMarkup Language (XML) Configuration Access Protocol (XCAP) allows aclient to read, write and modify application configuration data, storedin XML format on a server. XCAP maps XML document sub-trees and elementattributes to HTTP URIs, so that these components can be directlyaccessed by HTTP. Additional details of XCAP is provided in “TheExtensible Markup Language (XML) Configuration Access Protocol (XCAP)”by J. Rosenberg, Internet Draft, IETF work in progress, July 2004 (whichis incorporated herein by reference in its entirety).

XML Configuration Access Protocol (XCAP) allows a client to read, writeand modify application configuration data, stored in XML format on aserver. The data has no expiration time, so it must be explicitlyinserted and deleted. The protocol allows multiple clients to manipulatethe data, provided that they are authorized to do so. XCAP is used inSIMPLE based presence systems for manipulation of presence lists andpresence authorization policies. Thus, XCAP is rather suitable forproviding device independent presence document manipulation.

A series of related textual messages between two or more parties can beviewed as part of a session with a definite start and end. This is incontrast to individual messages each sent completely independently.Under the SIMPLE standards, messaging schemes only track individualmessages as “page-mode” messages, whereas messaging that is part of a“session” with a definite start and end is called “session-mode”messaging.

Page-mode messaging is enabled in SIMPLE via the SIP MESSAGE method.Session-mode messaging has a number of benefits over page-mode messaginghowever, such as explicit rendezvous, tighter integration with othermedia types, direct client-to-client operation, and brokered privacy andsecurity.

The Contact Information for Presence Information Data Format (CIPID) isan extension that adds elements to PIDF that provide additional contactinformation about a presentity and its contacts, including references toaddress book entries and icons. CIPID is further detailed in “CIPID:Contact Information in Presence Information Data Format” by H.Schulzrinne, Internet Draft, IETF work in progress, July 2004 (which isincorporated herein by reference in its entirety).

Presence information, e.g., represented as Presence Information DataFormat (PIDF) and Rich Presence Information Data Format (RPID) describesthe current state of the presentity. RPID also allows a presentity toindicate how long certain aspects of the status have been valid and howlong they are expected to be valid, but the time range has to includethe time when the presence information is published and delivered to thewatcher. This restriction is necessary to avoid backwards-compatibilityproblems with plain PIDF implementations. RPID is additionally describedin “RPID: Rich Presence Extensions to the Presence Information DataFormat” by H. Schulzrinne et al., Internet Draft, IETF work in progress,March 2004 (which is incorporated herein by reference in its entirety).Likewise, PIDF is further detailed in “Timed Presence Extensions to thePresence Information Data Format (PIDF) to Indicate Presence Informationfor Past and Future Time Intervals” by H. Schulzrinne, Internet Draft,IETF work in progress, July 2004 (which is incorporated herein byreference in its entirety).

In some cases, the watcher can better plan communications if it knowsabout the presentity future plans. For example, if a watcher knows thatthe presentity is about to travel, it might place a phone call earlier.

It can also be useful to represent past information as it may be theonly known presence information. Such past information may providewatchers with an indication of the current status. For example,indicating that the presentity was at a meeting that ended an hour agoindicates that the presentity is likely in transit at the current time.

FIG. 8 shows exemplary multimodal wireless and wired devices that canaccess a variety of disparate networks using pertinent communicationstacks and physical network ports to those networks. According tovarious embodiments of the present invention, multimodal communicationdevices 801 a-801 d can have mobile phone capabilities as well ascomputing functions (e.g., Personal Digital Assistant (PDA)). Theseexemplary devices 801 a-801 d can provide PC-phone/PDA applications, PDAsynchronization, “dial” from the PC, etc. The device 801 c, forinstance, can include a Wi-Fi™ terminal for use in the office or homenetwork, and can also be a desktop speakerphone having a suitabledesktop socket. By way of example, suitable sockets for the multimodalcommunication devices 801 a-801 d have one or more of the followingfunctions: battery charger, PC/laptop synchronization, Ethernet RJ-45jack, a speaker (e.g., for quality room speakerphone), and a colordisplay for presence and IM without the PC/laptop.

The multimodal communication devices 801 a-801 d can also be a wired orwireless IP Centrex like phone with applications beyond voice—e.g., suchas presence, events, IM, conferencing collaboration and games. As noted,these devices 801 a-801 d can assume the role of a PBX or can interworkwith existing PBXs.

These multimodal devices 801 a-801 d advantageously provide users withenhanced capability over traditional stations, primarily because thesedevices 801 a-801 d can store and/or execute valuable data andsophisticated applications, such as personal data (e.g., address bookand calendar), various office applications, entertainment (e.g., musicand video files), account information for various services includingconverged communications, and payment mechanisms, etc.

A multimodal communication device (e.g., 801 a-801 d) can containsoftware stacks 803 and 805 for mobile networks (e.g., 2G and 3G, etc.)and for Internet access using Wi-Fi™/WiMax and wired Ethernet LANs.Accordingly, the lower stack 803 includes Layer 1 (L1) and Layer 2 (L2)protocols, while the upper stack 805 can include User Datagram Protocol(UDP), Transmission Control Protocol (TCP) and Internet Protocol (IP),as well as G2.

As shown, gateways 807 are utilized to provide seamless communicationsto the respective networks: PSTN 807, cellular networks 809 and 811(e.g., 2G, Code Division Multiple Access (CDMA), Global System forMobile Communications (GSM), etc.), and the Internet 813. For example,the 2G network 809 (CDMA and GSM) may support only voice and SMS, whilethe 3G network 811 may provide 3GPP IMS (3rd Generation PartnershipProject IP Multimedia Subsystem) services.

In an exemplary embodiment, some of the functions described can beaccomplished using a Bluetooth link between the multimodal communicationdevice (e.g., 801 a-801 c) and the PC/laptop or with a Bluetooth enabledSIP phone that is connected to the Internet 813—notably for suchfunctions as ICE and STUN/TURN servers for NAT and firewall traversal.

The following process describes network and service access to Internetbased SIP services by the multimodal devices 801 a-801 d. First, an IPaddress is obtained, for example, using Dynamic Host ConfigurationProtocol (DHCP). Thereafter, Internet access is achieved. ICE providesdetermination of the optimum NAT/firewall traversal. The device 801 a,for instance, can then register with the home SIP registrar to receivethe SIP based IP communication services. According to one embodiment ofthe present invention, a SIP re-INVITE is utilized to switch betweennetworks without leaving an established session, such as a conference.

Smooth handoff in wireless networks can be readily accomplished at theNetwork Layer 2, in the respective radio networks, such as in 2G/3G orWi-Fi™/WiMax networks. The user may be prompted by the mobile device 801a to approve the switch from one network type to another, such as whenswitching from the mobile 2G network 809 to an enterprise or hot spotWi-Fi™ network (not shown). In contrast to approaches where both avisited SIP registrar and a home SIP registrar are utilized, the systemcan utilize a single SIP registrar (e.g., the home registrar).

It is also contemplated that similar techniques may be applied forallowing a user to move from one device/interface to another whilemaintaining a given session.

As seen in FIG. 8B, the multimodal mobile device 801 (of FIG. 8A)includes a cellular transceiver 851 for communication with cellularsystems. A wireless transceiver 853 is also included for connecting towireless networks (e.g., 802.11, etc.). Further, a network interfacecard (NIC) 855 is provided for connectivity to a wired network; the NIC855 can be an Ethernet-type card. Use of the transceivers 853, 855 orNIC 855 depends on the mode of operation of the device 801, and iscontrolled by a controller 857. Radio transmissions can be relayed viathe antenna 861.

The multimodal mobile device 801 additionally includes a processor 863for executing instructions associated with the various applications(e.g., PDA functions and applications, etc.), as well as memory 865(both volatile and non-volatile) for storing the instructions and anynecessary data.

FIGS. 9-15 are diagrams of various call flows involving the multimodaldevices. For the purposes of explanation, these processes are describedwith respect to the system 700 of FIG. 7.

FIG. 9 is a diagram of a process for authentication and registration ofa multimodal device in a data network, according to one embodiment ofthe present invention. In step 901, the mobile station 801 connects tothe Access Point 713 (which in this example is an 802.1 accesspoint/Ethernet switch) using an Extensible Authentication Protocol(EAP). The Access Point 713 then communicates using EAP over RADIUS, asin step 903, with the AAA server 715. This server 715 is considered a“visited” RADIUS AAA server 715. The AAA server 715 then issues aRequest message for authentication to the AAA server 717 of the serviceprovider network 709 (step 905). The AAA server 717 responds with anAnswer message, per step 907. In turn, the Visited AAA server 715returns a Response message to the Access Point 713, which signals an EAPSuccess to the mobile station 701, per steps 909 and 911.

In step 913, the mobile station 701 and the Access Point 713 perform aDynamic Host Configuration Protocol (DHCP) process. Next, the mobilestation 701 establishes communication with the STUN/TURN server 719, asin step 915. Thereafter, communication with the SIP server 721 isexecuted by the mobile station 701 through a REGISTER and 200 OKexchange, per steps 917 and 919.

FIG. 10 is a diagram of a process for establishing a call from amultimodal device to the PSTN, according to one embodiment of thepresent invention. By way of example, this call flow is performed incellular (e.g., 2G) mode, whereby the mobile station 701 performs a callattempt specifying the dialed digits to the cellular mobile switch 723(step 1001). In step 1003, the cellular mobile switch 723 signals a callsetup request (ISUP Initial Address Message (IAM) or Setup with dialeddigits) to the mobile gateway 725. The gateway 725 then generates anINVITE message to the SIP proxy server 721, per step 1005. The server721 conveys the INVITE to the PSTN gateway 727, which responds with a200 OK message (steps 1007 and 1009).

The SIP proxy server 721 forwards, as in step 1011, to the mobilegateway 725. This gateway 725 consequently sends, per step 1013, anAnswer Message (ANM) or Connect message to the cellular mobile switch723. In step 1015, the switch 723 signals a Connected message to themobile station 701.

Per step 1019, the mobile station 701 and a phone off the PSTN can begincommunicating as a call is now established.

The above call flow involves a call being initiated by the mobilestation 701; the following process describes a call being received bythe mobile station 701 from a station within the PSTN 711.

FIG. 11 is a diagram of a process for establishing a call to amultimodal device from the PSTN, according to one embodiment of thepresent invention. In this scenario, a station within the PSTN 711places a call to the mobile station 701. The PSTN gateway 727 sends anINVITE message, per step 1101, to the SIP proxy server 721, whichforwards the INVITE message to the mobile gateway 725 (step 1103). Instep 1105, the mobile gateway 725 sends an IAM or Setup message to thecellular mobile switch 723. The switch 723 then signals an Alertingmessage to the mobile station 701, per step 1107. In step 1109, themobile station 701 responds with an Answer to the cellular mobile switch723. The switch 723 next relays an ANM or Connect message, as in step1111, to the mobile gateway 725.

In response to the Connect message, the mobile gateway 725 transmits a200 OK message to the SIP proxy server 721 (step 1113). This server 721subsequently forwards the 200 OK message to the PSTN gateway 727, perstep 1115. In step 1117, the PSTN gateway 727 replies with an ACKmessage to the SIP proxy server 721, which relays this message to themobile station 725 (step 1119). Thereafter, a call is establishedbetween the mobile station 701 and the originating station, as in step1121.

FIG. 12 is a diagram of a process for cellular-to-IP mode switchingduring a call supported by the PSTN, according to one embodiment of thepresent invention. It is assumed that a cellular call (in 2G) is inprogress (step 1201). In step 1203, the mobile station 701 authenticateswith the Access Point 713. Also, the mobile station 701 performs SIPregistration (STUN/TURN) via the SIP proxy server 721, per step 1205.Next, the mobile station 701 sends an INVITE message to the SIP proxyserver 721, which communicates with the PSTN gateway 727 (steps 1207 and1209). The PSTN gateway 727 replies with a 200 OK message, per step1211; the gateway 727 forwards the 200 OK message to the mobile station701 (step 1213).

After receiving the 200 OK message, the mobile station 701 replies, asin step 1215, to the SIP proxy server 721 with an ACK message. Per step1217, the SIP proxy server 721 transmits the ACK message to the PSTNgateway 727.

At this stage, the PSTN gateway 727 signals the termination of the 2Gcall with a BYE message to the SIP proxy server 721, per step 1219. Theproxy server 721 forwards the BYE message to the mobile gateway 725, asin step 1221. In step 1223, the mobile gateway 725 sends a Releasemessage to the cellular mobile switch 723, which sends a Disconnectmessage to the mobile station 701.

After sending the Release signal, the mobile gateway 725 also sends a200 OK message, as in step 1227, to the SIP proxy server 721. The proxyserver 721 sends the 200 OK message to the PSTN gateway 727. Therefore,an IP call is established, per step 1231.

Alternatively, the mobile station 701 can switch from an IP call to a 2Gcall, as next explained.

FIG. 13 is a diagram of a process for IP-to-cellular mode switchingduring a call supported by the PSTN, according to one embodiment of thepresent invention. In step 1301, the mobile station 701 has establisheda packetized voice call (e.g., operating in IP mode) with a stationwithin the PSTN 727. The mobile station 701 sends a call attemptrequest, which indicates the dialed digits to the cellular mobile switch723 (step 1303). The cellular mobile switch 723 sends a call setuprequest, IAM or Setup with dialed digits, to the mobile gateway 725, perstep 1305. The mobile gateway 725 generates an INVITE message to the SIPproxy server 721, per step 1307. The server 721 sends the INVITE to thePSTN gateway 727 (step 1309), which responds with a 200 OK message (step1311). The proxy server 721 sends the 200 OK message to the mobilegateway 725, as in step 1313.

In step 1315, the mobile gateway 725 sends an ANM (Answer Message) orConnect message to the cellular mobile switch 723. The switch 723signals a Connected message to the mobile station 701, per step 1317.

The mobile gateway 725 sends an ACK message, per step 1319, to the SIPproxy server 721, which transmits the ACK message to the PSTN gateway727 (step 1321). Thereafter, the PSTN gateway 727 sends a BYE message tothe SIP proxy server 721, which forwards the message to the mobilestation 701 (steps 1323 and 1325). In step 1327, the mobile station 701transmits a 200 OK message to the SIP proxy server 721; the 200 OKmessage is further sent to the PSTN gateway 727 (step 1329).Consequently, a TDM call is now supported between the mobile station 701and the PSTN station.

FIG. 14 is a diagram of a process for call establishment by a multimodaldevice operating in cellular mode, according to one embodiment of thepresent invention. Under this scenario, two mobile stations A and B areinvolved in the call flow. The mobile station A signals a call attemptwith the cellular mobile switch 723 (step 1401). The cellular mobileswitch 723 sends an IAM or Setup message to the mobile gateway 725, perstep 1403. The mobile gateway 725 generates an INVITE message to the SIPproxy server 721, per step 1405.

In step 1407, the SIP proxy server 721 to the mobile gateway 725, whichtransmits an ISUP (ISDN User Part) Initial Address Message (IAM) orSetup message to the cellular mobile switch 723 (step 1409). Thecellular mobile switch 723 exchanges Alerting/Answer signaling withmobile station B, per step 1411. The cellular mobile switch 723 sends anANM or Connect message to the mobile gateway 725 (step 1413). Next, themobile gateway 725 generates, as in step 1415, a 200 OK message to theSIP proxy server 721. The proxy server 721 responds back with a 200 OKmessage, per step 1417.

In step 1419, the mobile gateway 725 sends an ANM or Connect message tocellular mobile switch 723. A connection is established with the mobilestation A (step 1421).

Per step 1423, the mobile gateway 725 sends an ACK message to the SIPproxy server 721, which transmits its own ACK message to the mobilegateway 725 (step 1425). Hence, the cellular mobile switch 723 hasestablished cellular communication with both the mobile stations A andB, per steps 1427 and 1429.

FIG. 15 is a diagram of a process for cellular-to-IP mode switchingmid-call, according to one embodiment of the present invention. Thisscenario involves a cellular call being in progress between the mobilestation A and the mobile station B, as in steps 1501 and 1503. In step1505, the mobile station A performs an 802.1 authentication with theAccess Point 723. Also, the mobile station A performs SIP registrationwith the STUN/TURN functions via the SIP proxy server 721 (step 1507).In step 1509, the mobile station A sends an INVITE message to the SIPproxy server 721. The SIP proxy server 721 then sends an INVITE messageto the mobile gateway 725, per step 1511. The mobile gateway 721generates a 200 OK message to the SIP proxy server 721, which sends the200 OK message to the mobile station A (steps 1513 and 1515).

In step 1517, the mobile station A forwards an ACK message to the SIPproxy server 721 in response to the 200 OK message. The SIP proxy server721, per step 1519, sends an ACK to the mobile gateway 725. The mobilegateway 725 next sends a BYE message to the SIP proxy server 721 (step1521).

The mobile gateway 725 next sends a Release message to the cellularmobile switch 723, which in turn issues a Disconnect message to themobile station A (steps 1523 and 1525).

The SIP proxy server 721, in step 1527, transmits a BYE message to themobile gateway 725, which responds with a 200 OK message (steps 1527 and1529). At this point, the mobile station B still engaged in a cellularcall leg, per step 1531. In step 1533, the SIP proxy server 721 sends a200 OK to the mobile gateway 725. Now, the mobile station Acommunicating over IP media, as in step 1535.

FIG. 16 is a diagram of an Operational Support System (OSS)architecture, according to one embodiment of the present invention. Thearchitecture 1600 leverages service-oriented architecture principles andassociated technologies. For example, remotely callable services,implemented using Web Services standards, are used to encapsulate accessto databases; encapsulate access to existing or “legacy” systems (asnecessary). These services advantageously provide OSS functionimplementations that are modular. Additionally, the callable servicesprovide interfaces for other systems to send notifications to IP-ICcomponents and to request information. These services furtheradvantageously provide a clean, platform-agnostic, standards-baseddecoupling between web-facing and back-end systems.

According to one embodiment of the present invention, the architecture1600 includes three primary tiers: an Access Tier 1601, a Services Tier1603, and a Resource Tier 1605. The Access Tier 1601 (which can also bereferred to as a “Presentation Tier”) permits user and system accessinto the OSS for customers and service provider's sales/support. TheServices Tier 1603 is the focal point of the OSS architecture 1600,where a majority of the functionalities reside. Lastly, the ResourceTier 1605 encompasses the elements that the services act upon. The OSSarchitecture 1600 manages these various resources.

According to one embodiment of the present invention, the subsystems ofthe Access Tier 1601 include a Web Portal 1607, a Web Services Gateway1609, and an Identity Management and Access Control component (notshown). These interrelated components allow human users (e.g., customeremployees or service provider's staff) and customer systems 1611 toaccess the OSS services via, for example, web browser 1611 or via SimpleObject Access Protocol (SOAP) invocations.

In an exemplary embodiment, the external access architecture are asfollows. A web server is provided in a DMZ. Also, programming andruntime environment is supported for dynamic generation of HTML pagesand for handling incoming web requests. An XML firewall is deployed forscreening and routing inbound SOAP traffic coming into DMZ fromcustomers. Also, by way of example, web server agents are plugged intothe web server and XML firewall. Further, a Policy Server and LDAPbacking store can be utilized.

The identity administration allows authorized users to be added, and topermit these users to enter orders, update information, provision users,etc., on behalf of their organization or company. This administrationfunction enable delegation of administration privileges to customeradministrators, allowing them to add further users and grant them accessprivileges. It is assumed the service provider has some control inidentity administration, as the customer cannot be completelyself-managed using, e.g., web self-service. It is important to note thatthis identity administration function is distinct from end-user identitymanagement within the core SIP telephony components. The identityadministration is concerned with administrative accounts that allowcustomer employees to interact with the OSS systems online to allowcustomer self-service.

The Services Tier 1603 includes services that are mainly concerned withencapsulating resources, such as data and other managed resources,through Resource Encapsulation Services 1615. The Services Tier 1603also includes application process activities 1617—behavior, or actuallydoing something.

As shown, the arrows directed into the Services Tier 1603 constituteevent sources that trigger activities within the services. Exemplarytriggering events involve activities undertaken by the customer via webbrowser, notifications coming in from legacy systems (e.g., AccountsReceivable informing that a given customer has paid its bill), andmanagement-related notifications originating from IP Services componentsin the architecture. For example, a media relay server (or itsmanagement agent) can inform the OSS services that a resourceconsumption metric has gone above a high-water mark 1619 and additionalcapacity needs to be provisioned. Also, some OSS activities aretriggered by time-based events, as suggested by the hour glass. Inparticular, activities related to the monthly billing cycle are scheduledriven.

The Resource Tier 1605 includes databases 1621 and legacy systems 1623,as well as primary IP Services components 1625 (which are at the core ofthe IP-IC offering).

FIG. 17 is a diagram of a financial system for supporting the IPinterconnect service, according to one embodiment of the presentinvention. The system 1700 permits the IP-IC components to largelyperform their own billing computation and presentment and to integratewith existing financial systems 1701 (e.g., Accounts Receivable (AR) orother Finance systems). Alternatively, the system 1700 assume theintegration is a responsibility of these existing (or “legacy”)financial systems 1701. In either case, the system 1700 provides forencapsulating this integration point with a Web Service—this istransparent to the other components specific to the IP-IC OSS. Forexample, a clean SOAP interface to those existing systems is used, evenif that interface hides the legacy complexity of document file transferusing proprietary data formats.

As FIG. 17 shows, User Provisioning is invoked by the Access Tier 1601,driven by customer self-service events. The Access Tier 1601 then pushesupdates to the Customer Profile service and the ENUM/DNS servers. In oneembodiment, the system 1700 employs a GUI 1703, which provides one ormore Customer Self-Service screens to permit the user to provision andmanage their services. A Billing Presentment component 1705 is alsoprovided.

In an exemplary embodiment, presentment can be performed electronicallyvia the web portal 1607. The Billing Presentment component 1705 can bethough of as presentation code in the Web Portal 1607, which draws theunderlying statement information for each given customer from theBilling Statement store 1707, and renders that into, for example, HTMLmarkup for presentation to the user.

The User Provisioning component 1709, in an exemplary embodiment, is aWeb Service which provides interfaces for a single user, or a set ofmultiple users (possibly thousands), to be added to the system 1700. Theend-result of user provisioning, for instance, is that ENUM mappings forthe user(s), telephone number to SIP URI, are added to the ENUM DNSserver or servers 1710. Also, customer profile information is adjustedto increment or decrement the current user count field for the customeror customers. According to one embodiment of the present invention,mirror databases are updated with the ENUM mapping information. Thisinformation can be captured in database format (in addition to DNS) forother uses, e.g., to support white pages directory.

Because the User Provisioning component is implemented as a Web Service,the Application Programming Interface (API) can include methods foradding a single user to the system, dropping a single user from thesystem, bulk-loading an array of users to the system, and for performingbulk drops. These API functions can be exposed to the customers as XMLWeb Services interfaces, which the customer systems 1613 canprogrammatically call. The customer self-service screens of the IP-ICWeb Portal can also provide Graphical User Interface (GUI) interfacesallowing customer administrative personnel to add and drop users.

Additionally, the User Provisioning component 1709, according to oneembodiment of the present invention, performs dynamic updates to the DNSserver or servers. By way of example, the dynamic update can be executedby using public domain Java™ APIs into DNS, using available C languagelibrary and use JNI to support binding of Java™ code to object code, orexercise available DNS management interfaces. In an exemplaryembodiment, one of the roles of the User Provisioning service is to hidethe exact details of this DNS binding from upstream systems, so allthese upstream systems “see” a simple Web Service interface.

When the User Provisioning component 1709 adds or drops a user (orusers) for a given Customer, the Customer Profile service 1711 updatesbookkeeping on the user count. This can include updating a current usercount field and updating a monthly peak user count field with respect tothe User Provisioning component 1709. The Customer Profile component1711 also interacts with a Billing Computation component 1713 and aFulfillment (also referred to as an Order Management/CustomerProvisioning) component 1715.

Within the IP-IC service, the notion of provisioning can occur, in anexemplary embodiment, at two different levels: (1) provisioning andde-provisioning of individual SIP end-users (an ongoing activity), and(2) provisioning of customers. In contrast with up-front activities ofprovisioning a new customer, configuring a given customer facility orPBX to point to IP-IC DNS, redirect, relay and/or signaling conversionservers, etc. The User Provisioning service 1709 described in thisexample focuses on the former notion of provisioning the SIP end-user,not customer-level provisioning. The Fulfillment component 1715 focuseson the customer-level sense of provisioning.

According to an embodiment of the present invention, the BillingComputation component (or engine) 1713 is a service that is primarilyprocess-oriented. It is triggered by a scheduler 1717—e.g., on a monthlybilling cycle. Depending upon the service pricing model, the BillingComputation component 1713 can also be triggered on a daily basis inorder to take a daily sample of each Customer's user count. The samplescan then be used to update a running accumulator for the purpose ofcalculating a monthly average user count, for instance.

As for the Rating component 1719, this function can be integrated intothe billing computation, with regard to applying relevant discounts.

For the purposes of illustration, it is assumed that the pricing modelis based upon peak user count over the course of the month, rather thanthe average. As discussed above, the peak user count is maintained bythe Customer Profile component 1711, each time it gets anincrement/decrement user count event from the User Provisioning Service1709. On a monthly trigger event, the Billing Computation engine 1713cycles through the customers. The Customer Profile 1711 is queried forthe monthly peak user count for each customer. Each customer's ServiceProfile record 1721 is also consulted to determine the optional servicesthat the customer is subscribed to. The system 1700 allows for abusiness model where different features are optional, such as signalconversion or media relay, and such options incur additional chargesabove the base offering price.

Additionally, the Billing Computation engine 1713 pulls (and caches) thecurrent base price figures, for each option, from a Product Descriptionstore 1723. With all of this information, the Billing Computation engine1713 can then calculate the customer's itemized charges and bottom line.The Billing Computation engine can then consult the Rating component1719 to determine discount adjustments for the customer. Further, theBilling Computation engine 1713 prepares, for example, a XML documentthat represents the complete monthly information regarding what thecustomer bought and owes, and posts these XML documents to the BillingStatement store 1707. The Billing Statement 1707 store provides storageof these documents persistently for later consumption by the BillingPresentment component 1705 and financial systems 1701.

In an exemplary embodiment, the Billing Statement component 1707 is adata-oriented service, and supports persistent storage of the billingstatement documents that are created by the Billing Computation engine1713 for each customer (e.g., each month). Specifically, the BillingStatement component 1707 maintains storage for both the current billingcycle and for archival storage of all past billing statements.

In an exemplary embodiment, each record in the Billing Statement tablesstores an ASCII document. This document can be in XML format documentfor detailing the itemized charges for a given customer, applieddiscounts and bottom line. The XML document records the detail of whatthe customer bought, and what the customer owe. These XML documentsstored in the Billing Statement component 1707 represent all theinformation that is required for Billing Presentment 1705 to present ane-invoice to the customer, and for the financial systems 1701 to collectpayment and report back on the status of customer payment ordelinquency.

The Product Description component 1723 stores product informationreceived from the Product Design/Maintenance component 1725. In otherwords, the Product Description component 1723 is mainly a data store,and records information about the product offering as a whole, plusseparate information about each of the product's available options. Thisarrangement externalizes general information about the product so as toavoid hard-coding such information within program code. Of import ispricing information, which is likely subject to change, and best to keepin an external store. If a pricing model is adopted where separateproduct options are priced individually, then each option could have anassociated base price (or price rate per user).

The main client of the Product Description service 1723 is the BillingComputation engine 1713, which mainly needs to extract the base pricinginformation in order to compute bills.

The Service Profile component 1721 is another data-oriented component,and is fed by the Fulfillment component 1715 (which can be GUI driven byOrder Entry, Product Design and Customer Support web screens). TheService Profile component 1721 can be queried on a monthly cycle by theBilling Computation component 1713 in the course of calculating eachcustomer's bill.

The Service Profile component 1721 persists the complete productdescription, for each customer, of the products provisioned by thecustomer. If the product offering has several optional features (such assignal conversion, media relay, etc.), then the Service Profileinformation for each customer details the options elected by thecustomer, along with attributes that parameterize variable quantitiesassociated with the different product options. The Service Profilecomponent 1721 thus represents the instantiation of the IP-IC productoffering for each customer. This is in contrast with the ProductDescription component 1723, which embodies a description of the productas a whole, not any given customer's realization of the product. (Inobject-oriented parlance, the Product Description would be thought of as“class-level,” and the Service Profile would be “instance-level.”)

According to one embodiment of the present invention, the Fulfillmentcomponent 1715 provides a back-end to the customer self-service webscreens, as well as sales/support screens related to order managementand customer provisioning processes.

As noted earlier, provisioning involves multiple levels—provisioning inthe sense of enabling SIP end-users to use the system; and provisioningin the sense of “turning up” a new Customer and maintaining/updatingtheir information at a customer-level. The Fulfillment component 1715supports the customer-level sense of provisioning, not the SIP usermanagement, which is handled by the User Provisioning component.

Among other functions, the Fulfillment component 1715 supportsestablishing new customer accounts, and creating an IP-IC productspecific Accounts for an existing customer. In addition, the Fulfillmentcomponent 1715 can coordinate with customer data stores of record toensure that proper corporate Customer ID is used. The Fulfillmentcomponent 1715 also provides support for a customer entering survey oftheir needs and environment, which can assist sales personnel in productdesign/configuration. This Fulfillment component 1715 additionallyprovides Customer Premise Equipment (CPE) information entry, and caninform customers of the proper URLs or other binding information thatthey need for operational use of the various servers (e.g., DNS, ENUMRedirect, STUN, TURN, Signal Conversion gateways, etc.).

Moreover, the Fulfillment component 1715 permits customer election ofproduct options that define what the customer is buying. For example,the component can determine whether the customer require signalconversion, media relay, etc. Further, the Fulfillment component 1715supports entering site information.

As seen in the figure, the Fulfillment component 1715 communicates withan Inventory component 1727. In an exemplary embodiment, this Inventorycomponent 1727 is a data component that tracks relevant resourceinventory, both at the customer premises via the “legacy” customer datastore 1729 and resources that are internal to the service provider. Itis noted that separate stores for these two sorts of inventoryinformation can be maintained. For example, the inventory store can bekept in a relational database. By way of example, internal resourcesthat might be considered for storage in some sort of inventory serviceinclude CPUs (and their associated IP addresses), databases, deployedservices that comprise the OSS architecture. The inventory of deployedservices, according to an embodiment of the present invention, can bedeployed as a service directory, such as UDDI, rather than within arelational database. UDDI is a web-based distributed directory thatenables businesses to list themselves on the Internet.

FIG. 18 is a diagram of a service assurance infrastructure componentscapable of supporting the Interconnect services, in accordance with anembodiment of the present invention. The service assuranceinfrastructure 1800 can be thought of as a management plane (andsomewhat orthogonal to the other functional components discussedpreviously). Service assurance is a broad category of functions andsystems encompassing components and processes related to keeping thecore systems and support systems operational. Assurance functions caninclude monitoring, reporting, alarm management, capacity management andplanning, autonomic (self-healing) recovery techniques, Service LevelAgreement (SLA) management, policy-driven resource allocation, etc.

According to one embodiment of the present invention, it is assumed thatthe core of the service assurance architecture is based on aManager/Agent model. A number of different Agent types and instances(“active agents”) 1801 are responsible for monitoring the vital signs ofvarious resources 1803 (services, CPUs, databases) that make up thesystem environment. These active agents provide information to aManagement Layer 1805, which can be single tiered or multi-tiered. TheManagement Layer 1805 provides information to other interested systems,such as a management console 1807, capacity management component 1809,alerts 1811, and a report engine 1813, etc.

According to one embodiment of the present invention, the ManagementConsole 1807 can be a rich client. Such a rich client can be implementedwith Java™ applets, Java™ WebStart deployment of a Java™ application, ora .NET Smart Client, deployed perhaps with technology such as MicrosoftClickOnce technology (or via a hyper-link that resolves to an .exe,similar in spirit to the Java™ applet model).

The management infrastructure of the service assurance systemsdetermines when and where additional CPU resource are needed; alertscould be raised, and physical capacity could be provisioned (i.e.,another CPU rack installed). In light of these considerations, the Agenttier 1801 can be involved not only with monitoring health of deployedsystems, but also with dynamic deployment of services into theenvironment—service life-cycle management. For example, the growth ofthe core servers (e.g., Media Relay instances) supporting theInterconnect services can be readily management using the arrangement ofFIG. 18. The Media Relay instances can be deployed on-demand onto agrid-like farm of resources.

The processes described herein for supporting Interconnect services maybe implemented via software, hardware (e.g., general processor, DigitalSignal Processing (DSP) chip, an Application Specific Integrated Circuit(ASIC), Field Programmable Gate Arrays (FPGAs), etc.), firmware or acombination thereof. Such exemplary hardware for performing thedescribed functions is detailed below.

FIG. 19 illustrates a computer system 1900 upon which an embodimentaccording to the present invention can be implemented. For example, theprocesses described herein can be implemented using the computer system1900. The computer system 1900 includes a bus 1901 or othercommunication mechanism for communicating information and a processor1903 coupled to the bus 1901 for processing information. The computersystem 1900 also includes main memory 1905, such as a random accessmemory (RAM) or other dynamic storage device, coupled to the bus 1901for storing information and instructions to be executed by the processor1903. Main memory 1905 can also be used for storing temporary variablesor other intermediate information during execution of instructions bythe processor 1903. The computer system 1900 may further include a readonly memory (ROM) 1907 or other static storage device coupled to the bus1901 for storing static information and instructions for the processor1903. A storage device 1909, such as a magnetic disk or optical disk, iscoupled to the bus 1901 for persistently storing information andinstructions.

The computer system 1900 may be coupled via the bus 1901 to a display1911, such as a cathode ray tube (CRT), liquid crystal display, activematrix display, or plasma display, for displaying information to acomputer user. An input device 1913, such as a keyboard includingalphanumeric and other keys, is coupled to the bus 1901 forcommunicating information and command selections to the processor 1903.Another type of user input device is a cursor control 1915, such as amouse, a trackball, or cursor direction keys, for communicatingdirection information and command selections to the processor 1903 andfor controlling cursor movement on the display 1911.

According to one embodiment of the invention, the processes describedherein are performed by the computer system 1900, in response to theprocessor 1903 executing an arrangement of instructions contained inmain memory 1905. Such instructions can be read into main memory 1905from another computer-readable medium, such as the storage device 1909.Execution of the arrangement of instructions contained in main memory1905 causes the processor 1903 to perform the process steps describedherein. One or more processors in a multi-processing arrangement mayalso be employed to execute the instructions contained in main memory1905. In alternative embodiments, hard-wired circuitry may be used inplace of or in combination with software instructions to implement theembodiment of the present invention. Thus, embodiments of the presentinvention are not limited to any specific combination of hardwarecircuitry and software.

The computer system 1900 also includes a communication interface 1917coupled to bus 1901. The communication interface 1917 provides a two-waydata communication coupling to a network link 1919 connected to a localnetwork 1921. For example, the communication interface 1917 may be adigital subscriber line (DSL) card or modem, an integrated servicesdigital network (ISDN) card, a cable modem, a telephone modem, or anyother communication interface to provide a data communication connectionto a corresponding type of communication line. As another example,communication interface 1917 may be a local area network (LAN) card(e.g. for Ethernet™ or an Asynchronous Transfer Model (ATM) network) toprovide a data communication connection to a compatible LAN. Wirelesslinks can also be implemented. In any such implementation, communicationinterface 1917 sends and receives electrical, electromagnetic, oroptical signals that carry digital data streams representing varioustypes of information. Further, the communication interface 1917 caninclude peripheral interface devices, such as a Universal Serial Bus(USB) interface, a PCMCIA (Personal Computer Memory Card InternationalAssociation) interface, etc. Although a single communication interface1917 is depicted in FIG. 19, multiple communication interfaces can alsobe employed.

The network link 1919 typically provides data communication through oneor more networks to other data devices. For example, the network link1919 may provide a connection through local network 1921 to a hostcomputer 1923, which has connectivity to a network 1925 (e.g. a widearea network (WAN) or the global packet data communication network nowcommonly referred to as the “Internet”) or to data equipment operated bya service provider. The local network 1921 and the network 1925 both useelectrical, electromagnetic, or optical signals to convey informationand instructions. The signals through the various networks and thesignals on the network link 1919 and through the communication interface1917, which communicate digital data with the computer system 1900, areexemplary forms of carrier waves bearing the information andinstructions.

The computer system 1900 can send messages and receive data, includingprogram code, through the network(s), the network link 1919, and thecommunication interface 1917. In the Internet example, a server (notshown) might transmit requested code belonging to an application programfor implementing an embodiment of the present invention through thenetwork 1925, the local network 1921 and the communication interface1917. The processor 1903 may execute the transmitted code while beingreceived and/or store the code in the storage device 1909, or othernon-volatile storage for later execution. In this manner, the computersystem 1900 may obtain application code in the form of a carrier wave.

The term “computer-readable medium” as used herein refers to any mediumthat participates in providing instructions to the processor 1903 forexecution. Such a medium may take many forms, including but not limitedto non-volatile media, volatile media, and transmission media.Non-volatile media include, for example, optical or magnetic disks, suchas the storage device 1909. Volatile media include dynamic memory, suchas main memory 1905. Transmission media include coaxial cables, copperwire and fiber optics, including the wires that comprise the bus 1901.Transmission media can also take the form of acoustic, optical, orelectromagnetic waves, such as those generated during radio frequency(RF) and infrared (IR) data communications. Common forms ofcomputer-readable media include, for example, a floppy disk, a flexibledisk, hard disk, magnetic tape, any other magnetic medium, a CD-ROM,CDRW, DVD, any other optical medium, punch cards, paper tape, opticalmark sheets, any other physical medium with patterns of holes or otheroptically recognizable indicia, a RAM, a PROM, and EPROM, a FLASH-EPROM,any other memory chip or cartridge, a carrier wave, or any other mediumfrom which a computer can read.

Various forms of computer-readable media may be involved in providinginstructions to a processor for execution. For example, the instructionsfor carrying out at least part of the present invention may initially beborne on a magnetic disk of a remote computer. In such a scenario, theremote computer loads the instructions into main memory and sends theinstructions over a telephone line using a modem. A modem of a localcomputer system receives the data on the telephone line and uses aninfrared transmitter to convert the data to an infrared signal andtransmit the infrared signal to a portable computing device, such as apersonal digital assistant (PDA) or a laptop. An infrared detector onthe portable computing device receives the information and instructionsborne by the infrared signal and places the data on a bus. The busconveys the data to main memory, from which a processor retrieves andexecutes the instructions. The instructions received by main memory canoptionally be stored on storage device either before or after executionby processor.

While the present invention has been described in connection with anumber of embodiments and implementations, the present invention is notso limited but covers various obvious modifications and equivalentarrangements, which fall within the purview of the appended claims.

The following patent applications are incorporated herein by referencein their entireties: co-pending U.S. patent application Ser. No.11/202,659 filed Aug. 12, 2005, entitled “Method and System forProviding Voice Over IP Managed Services Utilizing a Centralized DataStore”; and co-pending U.S. patent application Ser. No. 11/202,589,filed Aug. 12, 2005, entitled “Fixed-Mobile Communications withMid-Session Mode Switching.”

APPENDIX

2G 2^(nd) Generation

3G 3^(rd) Generation

4G 4^(th) Generation

AAA Authentication,Authorization, and Accounting

ACK Acknowledgement

AIB Authenticated Identity Body

ALG Algorithm

ANM Answer Message

API Application Programming Interface

ASCII American Standard Code for Information Interchange

ASIC Application Specific Integrated Circuit

CA Certicificate Authority

CD Compact Disc

CIPID Contact Information for Presence Information Data Format

CPE Customer Premise Equipment

CPU Central Processing Unit

CRT Cathode Ray Tube

DSL Digital Subscriber Line

DDOS Distributed Denial of Service

DHCP Dynamic Host Configuration Protocol

DMZ Demilitarized Zone

DNS Domain Name Service/System

DVD Digital Versatile Disc (formerly Digital Video Disc)

EAP Extensible Authentication Protocol

ENUM Electronic Number

EPROM Erasable Programmable Read Only Memory

FPGA Field Programmable Gate Array

GUI Graphical User Interface

HTML HyperText Markup Language

HTTP HyperText Transfer Protocol

IAM ISUP Initial Address Message (IAM)

ICE Interactive Communication Establishment

IETF Internet Engineering Task Force

IIS Internet Information Services

IM Instant Messaging

IP Internet Protocol

IP-IC IP Interconnect

IPSec IP Security

IPTEL IP Telephony

ISDN Integrated Digital Services Network

ISP Internet Service Provider

ISUP ISDN User Part

ITU International Telecommunication Union

JNI Java™ Native Interface

LAN Local Area Network

LDAP Lightweight Directory Access Protocol

MAC Medium Access Control

MIME Multipurpose Internet Mail Extensions

NAPTR Naming Authority Pointer

NAT Network Address Translation

NIC Network Interface Card

OSS Operational Support System

PBX Private Branch Exchange

PCMCIA Personal Computer Memory Card International Association

PDA Personal Digital Assistant

PIDF Presence Information Data Format

POTS Plain Old Telephone Service

PROM Programmable Read Only Memory

PSTN Public Switched Telephone Network

RAM Random Access Memory

ROM Read Only Memory

RFC Request For Comment

RPID Rich Presence Information Data Format

RTP Real-Time Transport Protocol

RTCP Real-Time Control Protocol

SIMPLE SIP IM Protocols Leveraging Extentions

SIP Session Initiation Protocol

SIP-T Session Initiation Protocol for Telephones

SIPPING Session Initiation Proposal Investigation

S/MIME Secure/Multipurpose Internet Mail Extentions

SLA Service Level Agreement

SMS Short Messaging Systems

SOA IT Service Oriented Architecture Information Technology

SOAP Simple Object Access Protocol

SRTP Secure Real-Time Transport Protocol

STUN Simple Traversal of UDP

TCP Transmission Control Protocol

TDM Time Division Multiplexing

TLS Transport Layer Security

TURN Traversal Using Relay NAT

UA User Agent

UDDI Universal Description, Discovery and Integration

UDP User Datagram Protocol

URI Uniform Resource Indentifier

URL Uniform Resource Locator

VoIP Voice Over IP

VPN Virtual Private Network

WAN Wide Area Network

Wi-Fi™ Wireless Fidelity

WiMax Worldwide Interoperability for Microwave Access

WLAN Wireless Local Area Network

XCAP XML Configuration Access Protocol

XCON Centralized Conferencing

XML Extensible Markup Language

1. A method for providing communication services, the method comprising:receiving a request, at a proxy server, for establishing a voice callfrom a source endpoint behind a first network address translator of afirst domain, wherein the proxy server provides signaling forestablishment of the voice call and the request specifies a directorynumber of a destination endpoint within a second domain; convertingsignaling from another proxy server associated with establishment of thevoice call to a format compatible with the proxy server; selectivelydetermining a network address for communicating with the destinationendpoint based on the directory number; determining existence of asecond network address translator within the second domain; and if thenetwork address can be determined, establishing a media path between thesource endpoint and the destination endpoint based on the networkaddress to support the voice call.
 2. A method according to claim 1,wherein the step of determining the network address is performed by anENUM (Electronic Number) server, the step of supporting is performed bya STUN (Simple Traversal of UDP (User Datagram Protocol)) server, andthe step of establishing is performed by a TURN (Traversal Using RelayNAT (Network Address Translation)) server, each of the servers beingmanaged by a service provider.
 3. A method according to claim 2, whereinthe media path is established through the TURN server, and a proxyserver directs the voice call over the media path.
 4. A method accordingto claim 1, wherein the voice call is a Voice over IP (InternetProtocol) call.
 5. A method according to claim 1, wherein the firstdomain and the second domain correspond either to different enterprisenetworks or autonomous networks.
 6. A method according to claim 1,wherein the proxy server is located within a service provider network orthe first domain.
 7. A method according to claim 1, wherein the proxyserver utilizes Session Initiation Protocol (SIP) signaling, and theother proxy server utilizes either a SIP-type signaling or a H.323signaling.
 8. A method according to claim 1 further comprising: if thenetwork address cannot be determined, signaling a media gateway coupledto a circuit-switched telephone network for termination of the voicecall.
 9. A system for providing managed communication services, thesystem comprising: a proxy server configured to receive a request forestablishing a voice call from a source endpoint behind a first networkaddress translator of a first domain, wherein the request specifies adirectory number of a destination endpoint within a second domain; anaddress server configured to selectively determine a network address forcommunicating with the destination endpoint based on the directorynumber; a STUN (Simple Traversal of UDP (User Datagram Protocol)) serverconfigured to support determination of existence of a second networkaddress translator within the second domain; a TURN (Traversal UsingRelay NAT (Network Address Translation)) server configured to establish,if the network address can be determined, a media path between thesource endpoint and the destination endpoint based on the networkaddress to support the voice call, wherein, if the network addresscannot be determined, the proxy server communicates with a media gatewaycoupled to a circuit-switched telephone network for termination of thevoice call.
 10. A system according to claim 9, wherein the proxy serverdirects the voice call over the media path.
 11. A system according toclaim 9, wherein the voice call is a Voice over IP (Internet Protocol)call.
 12. A system according to claim 9, wherein the first domain andthe second domain correspond either to different enterprise networks orautonomous networks.
 13. A system according to claim 9, furthercomprising: a gateway configured to convert signaling from another proxyserver associated with establishment of the voice call to a formatcompatible with the proxy server.
 14. A system according to claim 9,wherein the proxy server utilizes Session Initiation Protocol (SIP)signaling, and the other proxy server utilizes either a SIP-typesignaling or a H.323 signaling.
 15. A system for providing communicationservices, the system comprising: means for receiving a request, at aproxy server, for establishing a voice call from a source endpointbehind a first network address translator of a first domain, wherein theproxy server provides signaling for establishment of the voice call andthe request specifies a directory number of a destination endpointwithin a second domain; means for converting signaling from anotherproxy server associated with establishment of the voice call to a formatcompatible with the proxy server; means for selectively determining anetwork address for communicating with the destination endpoint based onthe directory number; means for determining existence of a secondnetwork address translator within the second domain; and means forestablishing a media path between the source endpoint and thedestination endpoint based on the network address to support the voicecall, if the network address can be determined.
 16. A system accordingto claim 15, wherein the selectively determining means is an ENUM(Electronic Number) server, the determining means is a STUN (SimpleTraversal of UDP (User Datagram Protocol)) server, and the establishingmeans is a TURN (Traversal Using Relay NAT (Network AddressTranslation)) server, each of the servers being managed by a serviceprovider.
 17. A system according to claim 16, wherein the media path isestablished through the TURN server, and a proxy server directs thevoice call over the media path.
 18. A system according to claim 15,wherein the voice call is a Voice over IP (Internet Protocol) call. 19.A system according to claim 15, wherein the first domain and the seconddomain correspond either to different enterprise networks or autonomousnetworks.
 20. A system according to claim 15 wherein the proxy server islocated within a service provider network or the first domain.
 21. Asystem according to claim 15 wherein the proxy server utilizes SessionInitiation Protocol (SIP) signaling, and the other proxy server utilizeseither a SIP-type signaling or a H.323 signaling.
 22. A system accordingto claim 15, further comprising: means for signaling a media gatewaycoupled to a circuit-switched telephone network for termination of thevoice call, if the network address cannot be determined.